Index: webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc |
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc |
index 0711a133d4a9f5b49e5f43529803209e477779ca..3eb4eb63b3f3ee7c4ada0b87c7ebdbfbafe969fe 100644 |
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc |
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc |
@@ -46,7 +46,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, |
: timestamp_(0), |
packet_sent_(false), |
last_packet_send_timestamp_(timestamp_), |
- last_frame_type_(kFrameEmpty) { |
+ last_frame_type_(kSkipFrame) { |
AudioCoding::Config config; |
config.transport = this; |
acm_.reset(new AudioCodingImpl(config)); |
@@ -121,7 +121,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, |
const uint8_t* payload_data, |
size_t payload_len_bytes, |
const RTPFragmentationHeader* fragmentation) override { |
- if (frame_type == kFrameEmpty) |
+ if (frame_type == kSkipFrame) |
return 0; |
rtp_header_.header.payloadType = payload_type; |